[..] |
ASTERISK-10000: RTP Stream with wrong Timestamp after 200 ok when 183 session in progress |
ASTERISK-10001: chan_ooh323 don't work correctly |
ASTERISK-10002: No notyfications for Caller |
ASTERISK-10003: Display on 792x phones is incorrect for outbound calls that have Macros |
ASTERISK-10004: chan_iax2 tries to remove a nonexistend scheduled ping |
ASTERISK-10005: E&M Wink trucks no longer interpret dtmf digits correctly after 1.2.21.1 |
ASTERISK-10006: [patch] vnak_retransmit sends out incorrect frames |
ASTERISK-10007: "Mute" IAX2 Extensions |
ASTERISK-10008: ExternalIVR changes not playing audio |
ASTERISK-10009: Inbound FEATB calls not working with chan_zap on 1.4.8 |
ASTERISK-10010: [patch] Add manager command shell for accessing the system shell |
ASTERISK-10011: Seperate Full asterisk log files per day. |
ASTERISK-10012: app_dial segfaults asterisk while trying to bridge channels |
ASTERISK-10013: A rejection of an incoming call to a dynamic queue member causes the caller in the queue to be dropped |
ASTERISK-10014: chan_iax2 causes segfault while trying to queue a frame |
ASTERISK-10015: implementation of store_func & destroy_func |
ASTERISK-10016: Hint does not update state from Hold to Idle |
ASTERISK-10017: Asterisk locks and no new calls allowed. Doesn't crash but have to restart Asterisk to get calls to flow again. |
ASTERISK-10018: Crash in action_originate of manager.c |
ASTERISK-10019: [patch] implement skinny show settings |
ASTERISK-10020: If a local channel is a member of a queue, the queue will consider it invalid until at least one call has been made to it |
ASTERISK-10021: [patch] implement setvar functionality in chan_skinny |
ASTERISK-10022: externalivr "failed to execute" in recent versions; filename is not read correctly |
ASTERISK-10023: [patch] make cdr_pgsql.c compile in devmode |
ASTERISK-10024: wrong configure.ac check for gethostbyname_r on some platforms |
ASTERISK-10025: When using Asterisk RealTime, processing does not continue after queue command |
ASTERISK-10026: tab completion error asterisk CLI |
ASTERISK-10027: Put manage 'newexten' calls into new event |
ASTERISK-10028: Add a ring with annoucements option |
ASTERISK-10029: TE120XP |
ASTERISK-10030: Brand new bug: Asterisk dose not load information from mysql |
ASTERISK-10031: AST_LIST_REMOVE problems |
ASTERISK-10032: Asterisk Crashes in Function "retrans_pkt" |
ASTERISK-10033: [Patch] Incorrect logic when creating and queuing dynamic threads in socket_read() and iax2_process_thread() |
ASTERISK-10034: More doxygen changes mostly in res folder |
ASTERISK-10035: [patch] Restructure transmit_callstate function and calls to transmit_ |
ASTERISK-10036: Using Action: Monitor, if the File argument contain a dot (.), the call will never be mix |
ASTERISK-10037: hint stuck on hold when caller hangs up |
ASTERISK-10038: Repeated DTMF |
ASTERISK-10039: Stutter tone not produced on FXS module when new voice mail is present |
ASTERISK-10040: SIP_CODEC variable does not change the codec |
ASTERISK-10041: AGI timing affect pass-through faxing |
ASTERISK-10042: Application: SIPCallpickup for pickup up a call in the same pickup group |
ASTERISK-10043: [patch] Asterisk stops processing calls |
ASTERISK-10044: Support Diversion Header |
ASTERISK-10045: Support RFC 3325 |
ASTERISK-10046: CME causes crash during INVITE. |
ASTERISK-10047: Some sip ext go to "hold" and this ext don't receive calls |
ASTERISK-10048: Variables in AEL2 context names |
ASTERISK-10049: Transfer and One-Touch Recording not working |
ASTERISK-10050: [patch] Directory should list as many names as possible in one go |
ASTERISK-10051: One-way audio (one-way perfect and one-way distorted) |
ASTERISK-10052: [have fix] Transfers stopped working when migrating from 1.4.9 to 1.4.10 |
ASTERISK-10053: Makefile output expects 'make' as CC |
ASTERISK-10054: [patch] T.38 with devices behind NAT does not work in all circumstances |
ASTERISK-10055: MOH wierdness |
ASTERISK-10056: chan_sip.c HUGE bug |
ASTERISK-10057: Potential for DoS attack?: sip history recording can go on forever if SIP dialog never expires or is destroyed |
ASTERISK-10058: Unable to join queues until app_queue.so is reloaded |
ASTERISK-10059: All SIP peers stuck with Canreinvite=no |
ASTERISK-10060: Allow realtime members on non realtime queues/A member with priorty < 0 is logged out |
ASTERISK-10061: setting canrevinte=yes has no effect anymore on configuration of the peer |
ASTERISK-10062: WaitExten immediately hangs up channel |
ASTERISK-10063: inUse removed from users but remains for peers |
ASTERISK-10064: [patch] race condition in connection pool allocation |
ASTERISK-10065: Change 78181 breaks sip_notify |
ASTERISK-10066: Asterisk ignores port in register directive |
ASTERISK-10067: g726 appears broken between 1.4.x and 1.2.x |
ASTERISK-10068: app_record (and voicemail) don't record video to a file |
ASTERISK-10069: indications.c: Can't generate that much data! |
ASTERISK-10070: memory leak in zt_new() |
ASTERISK-10071: Weird output in "realtime mysql status" |
ASTERISK-10072: Memory leak and crash related (probably) to MixMonitor |
ASTERISK-10073: Polycom Phones Audio Bad (ala Mr Roboto) |
ASTERISK-10074: [patch] Asterisk segfaults after issuing mobile rfcomm without command |
ASTERISK-10075: One way totaly distorted audio, the other way is OK. |
ASTERISK-10076: billsec value changes depending on dialing method |
ASTERISK-10077: app_externalir is missing the E and V commands from external application |
ASTERISK-10078: [patch] implementation of store_func & destroy_func for SQL Lite |
ASTERISK-10079: [patch] implementation of store_func & destroy_func for ODBC |
ASTERISK-10080: Asterisk crashes when flooding with iax registrations and doing an "iax2 reload" or "reload" at the same time |
ASTERISK-10081: Maximum retries for seqno 102 when re-inviting. |
ASTERISK-10082: Sending caller ID to ZAP Extension fails if sendcalleridafter=0 or sendcalleridafter=1 |
ASTERISK-10083: Unavailable and Busy messages are deleted. |
ASTERISK-10084: Qualify intervals >1000ms create needless double OPTIONS transmissions |
ASTERISK-10085: Use SIP_TRANS_TIMEOUT instead of 32000 |
ASTERISK-10086: mISDN sporadically rejects incoming calls |
ASTERISK-10087: Ukrainian language in app_voicemal |
ASTERISK-10088: [patch] counter for iax2 show registry |
ASTERISK-10089: [patch] counter for voicemail show users |
ASTERISK-10090: No audio from/to Siemens S55 GSM phone |
ASTERISK-10091: [patch] renaming MOH functions not easily intelligible |
ASTERISK-10092: Extremely Choppy Calls after upgrade to 1.4.10.x |
ASTERISK-10093: Something in chanspy is crashing Asterisk |
ASTERISK-10094: [Patch] No ethernet interface found for seeding global EID .... always printed |
ASTERISK-10095: voicemail prompts can be truncated |
ASTERISK-10096: Prompts can not be interrupted with DTMF when using ZAP-Channels |
ASTERISK-10097: conf2ael improvements |
ASTERISK-10098: IAX2 crashes on trying to queue frames through the jb |
ASTERISK-10099: hint is hanging when remote party ends call on hold (re: 0010399 |
ASTERISK-10100: Missing documentation for %q option in logger.conf |
ASTERISK-10101: RFC2833 DTMF digits seem to be recognized, but the resulting sequence missing some digits |
ASTERISK-10102: MixMonitor Splits recordings |
ASTERISK-10103: Registration/Un-Registration Problem with chan_skinny |
ASTERISK-10104: Memory leak when ODBC used for voicemail storage |
ASTERISK-10105: SIP with canreinvite=yes through multiple Asterisk instances fails |
ASTERISK-10106: ztscan.conf unavailable |
ASTERISK-10107: [Patch] DUNDI calls app Dial with | seperators instead of , |
ASTERISK-10108: Calls Dial with the wrong parameter seperator |
ASTERISK-10109: callerid not passed to local channels |
ASTERISK-10110: [patch] Handle libcurl errors |
ASTERISK-10111: IMAP integration with MS Exchange crashes Asterisk-1.4.10.1 |
ASTERISK-10112: r79747 breaks config in at least chan_iax2.c and chan_sip.c |
ASTERISK-10113: Blind Transfer broken in 1.4.10.1 |
ASTERISK-10114: Memory leak in app_voicemail.c using IMAP backend |
ASTERISK-10115: zaptel-1.4 SVN, Makefile, Line 449, install-inlcude should read install-include |
ASTERISK-10116: Asterisk 1.4.9 and after crash when I check messages from peers behind NAT |
ASTERISK-10117: DTMF autogeneration (repeat last DTMF after change of line) |
ASTERISK-10118: chan_iax2.c seg fault in build_peer when "&context" is passed to strsep unitialised |
ASTERISK-10119: utils from trunk failing to build on OSX |
ASTERISK-10120: progress messages doesn't work |
ASTERISK-10121: pbx_ael crashs asterisk when unloading/loading |
ASTERISK-10122: [patch] update extentions.conf.sample to new style cli commands |
ASTERISK-10123: [patch] teach chan_iax2 to offer the calling channel's codec first, like chan_sip does it |
ASTERISK-10124: Transfer from macro hangs up before reaching the transfered context |
ASTERISK-10125: Check for callerid being present is incomplete. |
ASTERISK-10126: Callerid not working from pstn |
ASTERISK-10127: Patches to build V1.4.10.1 under Solaris 10 X86 |
ASTERISK-10128: audio is broken after 1 second |
ASTERISK-10129: Nonexistent extension makes crash asterisk |
ASTERISK-10130: Segfault when answering ringing mobile while monitoring call with mixmonitor |
ASTERISK-10131: channel appear as UP before call being picked up (and wrong indication in CDR) |
ASTERISK-10132: [patch] updates for cdr.conf.sample |
ASTERISK-10133: Gtalk no sound |
ASTERISK-10134: [patch] Output beautification for "[core] show codecs" |
ASTERISK-10135: [patch] misleading check for AUTH_UNKNOWN_DOMAIN |
ASTERISK-10136: the billsec is always 0 for SIP outgoing calls |
ASTERISK-10137: Asterisk crashed in function ast_dynamic_str_thread_build_va in utils.c |
ASTERISK-10138: [patch] Change chan_iax2.c to allow hosts to be poked if they have no ip and they do have dnsmgr |
ASTERISK-10139: [patch] install_prereq script for asterisk |
ASTERISK-10140: res_features CLI revamping and removal of default builtins |
ASTERISK-10141: Polycom Phones Audio Bad (ala Mr Roboto) post 1.4.9 |
ASTERISK-10142: ack announce cuts off audio file |
ASTERISK-10143: DTMFs passing only in part between two asterisk machines with an IAX2 connection |
ASTERISK-10144: [branch] bug in time-zone with daylight saving time |
ASTERISK-10145: Some information from INVITE of Peer A not passed to INVITE of Peer B |
ASTERISK-10146: [patch] For internal calls (e.g. Meetme) sse general language setting when it isn't set in peer definition |
ASTERISK-10147: /usr/src/asterisk/main/minimime/mm.h uint32_t not recognized |
ASTERISK-10148: [patch] DTMF detect debouncing logic can miss whole digits (dsp.c) |
ASTERISK-10149: When reloading moh and the directory is changed, the change doesn't apply |
ASTERISK-10150: Can't call skinny phone (Cisco 7960) |
ASTERISK-10151: Make Asterisk a Jabber to PSTN gateway |
ASTERISK-10152: Can't create applicationmap features with identical DTMF sequences-- even if only enable one per call |
ASTERISK-10153: timeout value should accept floating point numbers |
ASTERISK-10154: Queue members "disappear" when none are available. |
ASTERISK-10155: 2 crashes in chan_iax2 |
ASTERISK-10156: [patch] Some characters (such as #) are not escaped in SIP headers |
ASTERISK-10157: An incorrectly formatted voice mail will crash Asterisk |
ASTERISK-10158: [patch] Dundi support for true dynamic peering (the '*' mapping) |
ASTERISK-10159: cli command "gtalk show channels" output always "No gtalk channels in use" |
ASTERISK-10160: "Set" application silently changed in trunk to no longer support multiple assignments |
ASTERISK-10161: SIP Notify on Snom phones causes Asterisk to stop responding (not crash) |
ASTERISK-10162: IAX channel failure/crash in Trunk and Branch |
ASTERISK-10163: [patch] the fgets in res_agi can sometimes get interupted, but we dont handle it. |
ASTERISK-10164: Queue members on same IAX peer have same "in use" status |
ASTERISK-10165: Internal call to queue drops after exactly 1 minute |
ASTERISK-10166: Missing #ifdef PTRACING |
ASTERISK-10167: Show if meetme is locked |
ASTERISK-10168: ChannelReload manager event non-conformant - extra line break |
ASTERISK-10169: RTCP Stats show incorrect value for Our Sender Jitter: |
ASTERISK-10170: Crash in set_insecure_flags() function. |
ASTERISK-10171: [patch] no need to signal_condition after iax2_sched_add |
ASTERISK-10172: Registration failed with Wengo - Timeout |
ASTERISK-10173: Outbound Proxy Description |
ASTERISK-10174: DTMF INFO event appears to be causing Maximum retries exceeded on transmission hangup. |
ASTERISK-10175: Announce from queue does not seem to work. |
ASTERISK-10176: Doble inclusion of 'limitonpeer' parameter in sip.conf |
ASTERISK-10177: SIP hairpin invokes Local within app_dial to produce a crash. |
ASTERISK-10178: Cannot forward voicemail with vmail.cgi when voicemail is configured by res_config_mysql |
ASTERISK-10179: Bad hangup-cause in Hangup event: 16 instead 19, for calls originated using AMI |
ASTERISK-10180: Asterisk fails in Loop Back |
ASTERISK-10181: [patch] Free memory when pbx_dundi is unloaded |
ASTERISK-10182: Asterisk crashes on reload of pbx_config.so |
ASTERISK-10183: attended transfer with SIP channels: tranferer part doesn't see that called part answered, only with HAVE_EPOLL |
ASTERISK-10184: Attended transfer leaves two sides without audio path |
ASTERISK-10185: [patch] fix the redirect thru the ami |
ASTERISK-10186: Can not register with server from sip.conf if domain name or IP is not in real time db |
ASTERISK-10187: [RFC] export attended transfer functionality over AMI |
ASTERISK-10188: logger apparent deadlock |
ASTERISK-10189: [patch] RTP statistics returned by ast_rtp_get_quality reflects the last RTCP packet not a call overall |
ASTERISK-10190: [patch] changed a LOG_NOTICE to LOG_DEBUG |
ASTERISK-10191: Variables set in an IAX2 user/friend no longer make it to the channel created for an (authenticated?) incoming call |
ASTERISK-10192: crash on placing outbound call via SIP |
ASTERISK-10193: crash while send call via sip to another asterisk server into meetme |
ASTERISK-10194: SayNumber(0) doesnt work if format_gsm.so isnt loaded |
ASTERISK-10195: Asterisk Seg faults consistently at least once a day |
ASTERISK-10196: "logger rotate" leads to logging getting deadlocked and asterisk won't stop |
ASTERISK-10197: atxfer causes notice+warning |
ASTERISK-10198: Variable not defined in freepbx-cron-scheduler.php |
ASTERISK-10199: [patch] Failure reason missing in failed extension |
ASTERISK-10200: Codec options in gtalk.conf not respected |
ASTERISK-10201: 'Unknown' member status in app_queue |
ASTERISK-10202: [patch] updates for configuration samples |
ASTERISK-10203: Supervised Transfer gives continious tone in transferred party |
ASTERISK-10204: Wrong timestamp in IAX2 packets |
ASTERISK-10205: SIP/IAX trunks are being answered when being Queued. |
ASTERISK-10206: [patch] DNS SRV lookup behaviour is incorrect in some cases |
ASTERISK-10207: wrong error logged |
ASTERISK-10208: Unable to use variables containing times |
ASTERISK-10209: various coding errors causing more trouble than reasonable |
ASTERISK-10210: SIP or IAX2 ext. unavailable - chanisavail dies. If I unplug my Iaxy then calls don't go to my VM... |
ASTERISK-10211: Fix for #10599 breaks attended transfer |
ASTERISK-10212: [patch] Change verbose and debug levels to floating point |
ASTERISK-10213: [patch] re-order options |
ASTERISK-10214: [patch] menuselect: add page up/down support and visually indicate long category lists |
ASTERISK-10215: Application Pickup() "PICKUPMARK" string is not working! |
ASTERISK-10216: send_dtmf is call on chan_read() so theres no way to ignore dtmf in meetme |
ASTERISK-10217: iax2 craches on an empty host= line |
ASTERISK-10218: On incomming mobile-calls the asterisk crash |
ASTERISK-10219: svn trunk not compiling on centos 4.4 |
ASTERISK-10220: [patch] res_config_mysql.c wont compile |
ASTERISK-10221: Configuration file inclusion broken on reload |
ASTERISK-10222: One way sound on calls between mISDN and SIP |
ASTERISK-10223: [patch] fix memory leak with iax2 firmware when unloading chan_iax2 |
ASTERISK-10224: [Patch] updates to extensions.ael.sample for Voicemail syntax |
ASTERISK-10225: Followme attempts to open stream to no existing file |
ASTERISK-10226: new function EXTSTATE() returns state of an extension |
ASTERISK-10227: Meetme with Redirect leaves channel after hangup and crashes |
ASTERISK-10228: Zaptel unknown symbol |
ASTERISK-10229: Asterisk crash |
ASTERISK-10230: func_odbc call doesn't LOAD_FILE() |
ASTERISK-10231: [Patch] when using DESTDIR with make, update asterisk.conf too |
ASTERISK-10232: Config parser eats blank lines for dinner |
ASTERISK-10233: Config parser eats comments for breakfast |
ASTERISK-10234: [Patch] update followme.conf, was pointing to no existing prompt files |
ASTERISK-10235: Not a time watchdog to BYE request |
ASTERISK-10236: Extension status gets stuck on hold |
ASTERISK-10237: SIP Reinvite behaviour does not work as expected with certain dial() options |
ASTERISK-10238: followme.conf dialing channel |
ASTERISK-10239: RTP timestamp not updated |
ASTERISK-10240: Segfault inside spy_detach function |
ASTERISK-10241: Segmentation fault when chan_phone is unloaded is some cases |
ASTERISK-10242: advanced QUEUE_MEMBER_COUNT function |
ASTERISK-10243: SIP (re)INVITE after Channel Hangup |
ASTERISK-10244: [patch] Doxygen updates |
ASTERISK-10245: Queue with context requires multiple key presses to leave queue |
ASTERISK-10246: [patch] Do not log debug messages unless debug is enabled |
ASTERISK-10247: Memory leak if talk happend (1004 bytes for connection) |
ASTERISK-10248: CDRs are not merged for not answered calls |
ASTERISK-10249: [patch] SIP Session-Timers Support in Asterisk |
ASTERISK-10250: not optimal CDR lock flag check |
ASTERISK-10251: Audio problems (one direction) with 2 SIP peers when using queues and DTMF tones |
ASTERISK-10252: Incomplete CDR lock |
ASTERISK-10253: Multiple __SIPADDHEADERs don't work with Local channels |
ASTERISK-10254: crash in ast_obj2 (deletion) |
ASTERISK-10255: app_queue sets MEMBERINTERFACE just after it would be useful to use in MixMonitor file name variable substitution |
ASTERISK-10256: revision 81599 causes crash in gsm_read |
ASTERISK-10257: Compile failure in lock.h |
ASTERISK-10258: asterisk segfault in ast_list_traverse_safe_begin |
ASTERISK-10259: Wrong AST_STATE_UP state |
ASTERISK-10260: SQL 2005 Escape Character |
ASTERISK-10261: [post 1.4] SIP change at r77616 (rizzo) causes all outbound calls to fail authentication with 403 Forbidden |
ASTERISK-10262: Unnecessary options in misdn.conf |
ASTERISK-10263: [patch] gsm fails to compile with K6OPT (MMX) enabled and configure --enable-dev-mode |
ASTERISK-10264: memory leak due to ast_cdr_alloc invocations during call routing |
ASTERISK-10265: Handling of escaped characters (#, etc...) |
ASTERISK-10266: Crash Asterisk in the debug mode |
ASTERISK-10267: [patch] counter for database show |
ASTERISK-10268: [patch] avoid a bad ast_debug |
ASTERISK-10269: Wrong (???) user on incoming call |
ASTERISK-10270: [patch] change a log_notice to debug |
ASTERISK-10271: [patch] count the deltree entries on CLI datadata deltree foo |
ASTERISK-10272: [patch] A little code optimization |
ASTERISK-10273: [patch] Callback Application |
ASTERISK-10274: sip peer with missing close bracket causes all subsequent sip peers not to load |
ASTERISK-10275: WARNING[29872]: translate.c:163 framein: no samples for g729tolin |
ASTERISK-10276: [patch] Use of err() replace with fprintf solaris fix |
ASTERISK-10277: [patch] count numbers of users connected |
ASTERISK-10278: Doing an attended transfer, asterisk tries multiple times a native bridge |
ASTERISK-10279: [patch] autosupport script enhancements |
ASTERISK-10280: Systen Warnings |
ASTERISK-10281: [patch] Solaris build warnings |
ASTERISK-10282: [patch] asterisk -h cleanup |
ASTERISK-10283: [patch] Asterisk case sensitive problem. |
ASTERISK-10284: subscribecontext ignored |
ASTERISK-10285: look at http://bugs.digium.com/view.php?id=10546 |
ASTERISK-10286: Asterisk segfaulted using Mixmonitor |
ASTERISK-10287: Gtalk call fails unless restarts |
ASTERISK-10288: Followme system warnings |
ASTERISK-10289: asterisk segfaulted in __ast_module_user_remove |
ASTERISK-10290: Various cleanups |
ASTERISK-10291: At high load audio drops following - [Sep 13 09:55:35] DEBUG[10689] channel.c: Didn't get a frame from channel: Zap/1-1 |
ASTERISK-10292: [patch] CLI clear command |
ASTERISK-10293: Crash after second try to transfer, during atxfer |
ASTERISK-10294: [Sep 13 17:07:34] WARNING[29561]: rtp.c:887 ast_rtcp_read: RTCP Read too short |
ASTERISK-10295: typo in CODING-GUIDELINES |
ASTERISK-10296: ooh323 heap corruption every 3 days. |
ASTERISK-10297: The same noise problem from bluetooth to asterisk |
ASTERISK-10298: [patch] Code for USERUSERINFO support may create a global variable instead of a channel variable |
ASTERISK-10299: MemberName not reported in QueueMember and AgentCalled management events |
ASTERISK-10300: [patch] Allow odbc queries using SQLExecDirect() (ie. non prepared statement execution) |
ASTERISK-10301: [patch] Convert func_odbc to use SQLExecDirect() for speed |
ASTERISK-10302: [patch] use NEW_CLI for CLI commands. |
ASTERISK-10303: moh does not play during calls to func_curl |
ASTERISK-10304: moh does not play during calls to app_system |
ASTERISK-10305: "Presence" subscription causes Internal Server Error on Polycom 600 (re-open bug #5164) |
ASTERISK-10306: [patch] maxsecs option never set per mailbox due deprecated maxmessage, it's working globally only |
ASTERISK-10307: [patch] see channel in agi debug |
ASTERISK-10308: DISA prevents DTMF detection w/ FXO port on TDM400 |
ASTERISK-10309: Russell Janitor Project - NEW_CLI changes in pbx.c, logger.c and frame.c |
ASTERISK-10310: Russell Janitor Project - NEW_CLI changes in pbx.c, logger.c and frame.c |
ASTERISK-10311: res_features segmentation fault on Solaris 10 x86 |
ASTERISK-10312: [patch] Two missing manager command |
ASTERISK-10313: segfault when show dialplan displays res_features dislplan |
ASTERISK-10314: [patch] support for {cat,var}_metric and #include in res_config_sqlite |
ASTERISK-10315: [patch] Disable automatic move to Old from INBOX once 'heard' |
ASTERISK-10316: [patch] Move deleted messages to a Deleted Folder |
ASTERISK-10317: [patch] [depend:codec expansion] AMR NB pass-through |
ASTERISK-10318: Random replacement of channel name with other text in queue log entries |
ASTERISK-10319: Separate RTP pool for remote vs. local connections |
ASTERISK-10320: MobileStatus crash |
ASTERISK-10321: app_voicemail dumps core |
ASTERISK-10322: [patch] NoLossCDR work/cleanup/fixes tracking bug |
ASTERISK-10323: [patch] update http.conf.sample to reflect the deletion of cfgadvanced.html |
ASTERISK-10324: [patch] cdr_sqlite3_custom tries to load even when the config file isn't there |
ASTERISK-10325: While using queues, it crashes. Randomly. |
ASTERISK-10326: Duplicated and meaningless CDR Records |
ASTERISK-10327: macro defined in realtime that uses cmd MYSQL fails to connect |
ASTERISK-10328: Not supported option in sip.conf |
ASTERISK-10329: [patch] Finish reading extension after user pressed # |
ASTERISK-10330: [patch] updatecdr option for queues.conf |
ASTERISK-10331: NOTIFY contains invalid To header |
ASTERISK-10332: manager show connected - added output |
ASTERISK-10333: When the inbound call is not SIP, the outbound SIP call does not accepts setting the callerid via the CALLERID(num)= |
ASTERISK-10334: [patch] Extraneous </a> in Attached Files section |
ASTERISK-10335: Global subscribecontext setting is ignored |
ASTERISK-10336: [patch] incorrect CDR(lastapp) when monitor-type=MixMonitor |
ASTERISK-10337: Endless Loop occurs during 3 way call when callee disconnects. |
ASTERISK-10338: 'make asterisk.pdf' produces an unfound .sty error. |
ASTERISK-10339: Crash with "Segmentation fault" in zap internal timers processing |
ASTERISK-10340: Don't change dir to '/' on fork, dump the core right. |
ASTERISK-10341: Bad keepalive request from French FREE Provider with CIRPACK |
ASTERISK-10342: [patch] chan_sip may retransmit a SIP message after 2ms only |
ASTERISK-10343: [patch] Typo in debug traces |
ASTERISK-10344: NOTIFY messages are sent without content |
ASTERISK-10345: safe_asterisk incorrectly invokes /bin/sh instead of /bin/bash |
ASTERISK-10346: features don't work in AppDial |
ASTERISK-10347: [patch] Cannot compile on gcc-4.2 |
ASTERISK-10348: Asterisk suddenly slows down, and eats 100% cpu |
ASTERISK-10349: [patch] Portability fixes for openWRT |
ASTERISK-10350: Seg fault with ael contexts if empty |
ASTERISK-10351: An application send custom event through manager interface |
ASTERISK-10352: Call quality problem for IAX2 channel with G729 codec |
ASTERISK-10353: Segfaults on dial_exec_full() |
ASTERISK-10354: [patch] Doesn't lock config files when writing |
ASTERISK-10355: Video doesn't work for outgoing call? |
ASTERISK-10356: 0010754: [patch] Finish reading extension after user pressed # is not ok! |
ASTERISK-10357: [patch] ast_append_ha cleanup |
ASTERISK-10358: Allow assigning client status and priority |
ASTERISK-10359: #include directive does not work |
ASTERISK-10360: Impossible to make optional macro arguments in 1.4 |
ASTERISK-10361: for loops broken |
ASTERISK-10362: No way to set a variable without math |
ASTERISK-10363: Cannot add channel to group if using AMI Originate |
ASTERISK-10364: Latest svn update chan_zap won't compile |
ASTERISK-10365: [patch] MixMonitor called with old '|' separators |
ASTERISK-10366: WaitMusicOnHold() waits three times longer than it should. |
ASTERISK-10367: [patch] Add support for configuring a few SDP variables |
ASTERISK-10368: Segfault in chan_sip.c |
ASTERISK-10369: Incorrectly sized IAX2 frames sent from mISDN with b410p |
ASTERISK-10370: ASterisk Drops calls after 20 secs because of poor internet access |
ASTERISK-10371: ASterisk Drops calls after 20 secs because of poor internet access |
ASTERISK-10372: [patch] Allow ParkedCall to pickup the first parked call |
ASTERISK-10373: missing files in svn:ignore on utils/ |
ASTERISK-10374: Asterisk segfaults after an attended transfer to a queue using "Eyebeam" softphone. |
ASTERISK-10375: [patch] Add optional sendtext (like sendurl is now) |
ASTERISK-10376: Wrong definition of AST_KEY_DIR |
ASTERISK-10377: [patch] safe_asterisk includes bashisms |
ASTERISK-10378: Should be able to dynamically link against libc-client for IMAP storage |
ASTERISK-10379: [patch] make clean/distclean fixes |
ASTERISK-10380: [patch] SendFAX/ReceiveFAX |
ASTERISK-10381: errors on 'zap rsetart' |
ASTERISK-10382: [patch] app_queue: interface_exists_global not set return value... |
ASTERISK-10383: Hinting not working reliably |
ASTERISK-10384: menuselect fails to resolve dependencies |
ASTERISK-10385: [patch] Fixed missing MPEG4 Part 2 video codec pass-through in chan_sip.c |
ASTERISK-10386: Deffect in Attended Transfer code |
ASTERISK-10387: Add context field to app_meetme.so application for realtime mode |
ASTERISK-10388: PBX running Asterisk |
ASTERISK-10389: Poor audio quality and sometimes disconnection after hangup |
ASTERISK-10390: When originating a call from a meetme room, the ringback is not heard in the meetme room |
ASTERISK-10391: ChannelRedirect non-working |
ASTERISK-10392: [patch] useless buffer in zt_write() |
ASTERISK-10393: 302 Handling |
ASTERISK-10394: __zt_exception: We're Zap/1-1, not (null) |
ASTERISK-10395: IAX2: Out of idle IAX2 threads for I/O, pausing! |
ASTERISK-10396: [patch] asterisk crash while doing a 'module reload chan_sip.so' |
ASTERISK-10397: Wrong CDR |
ASTERISK-10398: [patch]Empty vars in AEL2 switches lead to dropped calls |
ASTERISK-10399: Add context field to app_meetme.so application for realtime mode |
ASTERISK-10400: Patch for an inteligent parkedcalls. |
ASTERISK-10401: SayNumber does not correctly pronounce numbers in dutch language |
ASTERISK-10402: Call parking crash |
ASTERISK-10403: usbradio.conf.sample should not be in 1.4 branch |
ASTERISK-10404: segmentation faults on installation with 3000 calls/day. |
ASTERISK-10405: Random crash in frame.c:328 |
ASTERISK-10406: AC_PREREQ should be 2.60 in configure.ac |
ASTERISK-10407: [JANITOR] Change simple snprintf to ast_copy_string |
ASTERISK-10408: [patch] preventing parallel logins from the same line or by the same agent |
ASTERISK-10409: AST_MODULE_INFO macro with wrong parameter order for c++ code modules.h |
ASTERISK-10410: executing CLI command from command line have bad verbosity |
ASTERISK-10411: Code cleanup in main/say.c |
ASTERISK-10412: [patch] app_queue tweak for QueueSummary manager event |
ASTERISK-10413: Manager Interface CDR Backend never being enabled |
ASTERISK-10414: Support for enabling and disabling particluar peer |
ASTERISK-10415: [patch] remove extra logging from ast_add_extension2 |
ASTERISK-10416: [patch] Sort music on hold files list |
ASTERISK-10417: [patch] Remove duplicate ast_cli_register from dnsmgr.c |
ASTERISK-10418: Voicemail to Email not working with ODBC_STORAGE |
ASTERISK-10419: Segfault if ast_cli_register is called multiple times with same data |
ASTERISK-10420: [patch] Fix randomness in musiconhold |
ASTERISK-10421: DUNDi use tos in deprecated format |
ASTERISK-10422: "H" and "h" option of dial command does not work if an extension uses explicit Answer |
ASTERISK-10423: Problem with DTMF being passed from Cisco GW to asterisk on ingress calls Description |
ASTERISK-10424: log complaining about tos setting |
ASTERISK-10425: Add --enable-dev-mode parameter to configure script |
ASTERISK-10426: chan_sip crashes on reload |
ASTERISK-10427: VoiceMailMain - Passcode check not skipped when passing s in the options |
ASTERISK-10428: [patch] New epoll API yields scrambled audio in one way |
ASTERISK-10429: 1.4.11 Stable - Polycom phones hang up when media is re-invited while resuming from an on-hold state |
ASTERISK-10430: Adding one touch parking to app_queue |
ASTERISK-10431: AEL & CUT |
ASTERISK-10432: [patch] Logging goes to wrong destination |
ASTERISK-10433: need to change the SQL Query |
ASTERISK-10434: parkingcalls, by context with minim code change and not config changes. |
ASTERISK-10435: Cannot install zaptel & asterisk packages |
ASTERISK-10436: Asterisk 1.4.12 crashes in channel.c |
ASTERISK-10437: [patch] Does not like , as seperator (still using |) |
ASTERISK-10438: [Patch] Fix handling RTP RFC2833 events > 16. They are not DTMF digits. |
ASTERISK-10439: app_voicemail does not obey anonymous caller id |
ASTERISK-10440: on "reload app_queue.so" Dynamic queue members are deleted |
ASTERISK-10441: Asterisk 1.4 console repeatedly showing *CLI> on MacOSX Tiger 10.4.10 |
ASTERISK-10442: Settings in sip.conf are not processed |
ASTERISK-10443: Remove .cvsignore file from the tree |
ASTERISK-10444: queue remove member <TAB> causes core dump and crash |
ASTERISK-10445: Queue recording via monitor-format and monitor-join produces output different from MixMonitor |
ASTERISK-10446: Crash in local_queue_frame trying to trylock a corrupted p->owner lock |
ASTERISK-10447: [patch] loge reason why DB can't be open |
ASTERISK-10448: [patch] Handle multiple commands as returned by the read system call |
ASTERISK-10449: When parking lot ring back times out, error is generated, line is hung up and timeout extension isn't reached. |
ASTERISK-10450: [patch] Add support for setting log levels on remote console |
ASTERISK-10451: Asterisk does not report time correctly |
ASTERISK-10452: Call Park from a queue not working |
ASTERISK-10453: pulsedial=no setting is runtime changed in chan_zap.c so you can't simply disable pulse dialling |
ASTERISK-10454: deadlock between ast_softhangup and zt_bridge (using mixmonitor on bridged channels) |
ASTERISK-10455: 2 features, ring expiry and periodic announce firstplay |
ASTERISK-10456: Parked call inherits transfer capability |
ASTERISK-10457: Enabling MTX Profile causes * not to compile |
ASTERISK-10458: 'reload app_queue.so' causes all queue members to disappear... |
ASTERISK-10459: [patch] AST_FORMAT_LIST incomplete for G.722 |
ASTERISK-10460: CUT() won't let you cut on a carriage return |
ASTERISK-10461: [patch] updates to sample zapata.conf |
ASTERISK-10462: [crash] asterisk segfaults using tab completion |
ASTERISK-10463: Change SQL order by clause that selects sections for better business-logic implementation |
ASTERISK-10464: "port" configuration parameter conflicts with runtime status |
ASTERISK-10465: Fix Round Robin Routing Allow 1 Up Port |
ASTERISK-10466: Bad /dev/zap/* permissions (zaptel.rules does not work) |
ASTERISK-10467: automon result in a continue dtmf tone |
ASTERISK-10468: VAD in meetme |
ASTERISK-10469: Delete from addons app_saycountpl |
ASTERISK-10470: Jabber gtalk not connecting anymore |
ASTERISK-10471: chan_cellphone has a NULL pointer dereference in the do_send_rfcomm routine |
ASTERISK-10472: Problems when doing an attended and unattended transfer with Thomson Phones |
ASTERISK-10473: Stop gracefully complains in _ast_pthread_mutex_unlock |
ASTERISK-10474: msg description file not closed properly after insufficiently long recording rejected |
ASTERISK-10475: Asterisk crashed by meetme |
ASTERISK-10476: [patch] Some double includes removed |
ASTERISK-10477: crash in ast_var_name on SIP hangup |
ASTERISK-10478: Support for Queued 182 messages |
ASTERISK-10479: Adding extra options to SIPPEER function. |
ASTERISK-10480: double call to ast_frame_free |
ASTERISK-10481: Zap assisted xfer via hookflash yield CDR errors and bad CDR's. |
ASTERISK-10482: E&M Wink stops responding after originating about 500 calls |
ASTERISK-10483: [queue] Allow setting port in external SIP member |
ASTERISK-10484: dialplan reload shoud also reload extensions.ael |
ASTERISK-10485: Added MySQL error message to connection errors |
ASTERISK-10486: Some of the last log messages are lost when terminating Asterisk |
ASTERISK-10487: [patch] Create a Dial option that switches from Ringing to Early Media |
ASTERISK-10488: No function for escaping shell characters |
ASTERISK-10489: Crash in ast_queue_frame |
ASTERISK-10490: app_mixmonitor crash in ast_channel_spy_read_frame |
ASTERISK-10491: make config does not honor DESTDIR |
ASTERISK-10492: stand alone dollar sign used in a string prevent normal variables substitution |
ASTERISK-10493: codec_g729a.so (v32) doesn't read licences file correctly |
ASTERISK-10494: [PATCH] Add a new queue strategy: weighted random (wrandom) |
ASTERISK-10495: [PATCH] Flag to disable Packet2packet bridging |
ASTERISK-10496: mISDN don't make hangup and crash |
ASTERISK-10497: [patch] delete non-existant schedule |
ASTERISK-10498: chan_sip can generate several outstanding requests, but ignores responses |
ASTERISK-10499: Content-Type: multipart/mixed not recognised correctly (SIP-T on not working) |
ASTERISK-10500: [patch] Division by zero error when a music on hold class is empty |
ASTERISK-10501: Compile asterisk-1.4.11 package error at funcs/func_curl.c |
ASTERISK-10502: [patch] Provides(fix) make rpm functionality to Asterisk |
ASTERISK-10503: [patch] Provide make rpm functionality to asterisk-addons |
ASTERISK-10504: SIP deadlocks unexpectedly at random intervals, trigger unknown |
ASTERISK-10505: srvlookup should be defaulted to "yes" in code and config example |
ASTERISK-10506: chan_vpb doesn't compile against latest stable release of Voicetronix drivers (4.2.18) |
ASTERISK-10507: 1.4.13 lockups |
ASTERISK-10508: chan_vpb sample configuration file messy, not well documented, and missing a few options |
ASTERISK-10509: [patch] use of unitialized buffer |
ASTERISK-10510: Spelling error SDMI listener |
ASTERISK-10511: channel.c:780 channel_find_locked // deadlocks. -URGENT |
ASTERISK-10512: [patch] Add HTTP Basic & Digest Auth (rfc2617) for manager web interface. |
ASTERISK-10513: LaTeX documentation update #3 |
ASTERISK-10514: SayNumber does not support billion |
ASTERISK-10515: Directory doesn't work |
ASTERISK-10516: compilation fails at cli.c |
ASTERISK-10517: [patch] sample extensions.ael uses deprecated feature |
ASTERISK-10518: Wait() casuses segfault |
ASTERISK-10519: Remote console command completion is broken |
ASTERISK-10520: [patch] default context is created even if users.conf does not have users |
ASTERISK-10521: [patch] safe/limited Originate manager action |
ASTERISK-10522: several crashes in Trunk using cdr_odbc and func_odbc with SQL Server. fretds 0.64 (latest) |
ASTERISK-10523: No way to exit from background() application... |
ASTERISK-10524: CLI autocompletion stopped working |
ASTERISK-10525: NAT settings ignored on calls recieved to [general] |
ASTERISK-10526: [patch] multiple buffer problems with DUNDi |
ASTERISK-10527: [patch] LaTeX documentation update #4 |
ASTERISK-10528: [path] main/util.c missed *dst='\0'; in ast_base64decode() |
ASTERISK-10529: [patch] Do not log debug messages unless debug is enabled |
ASTERISK-10530: Daylight Time Saving |
ASTERISK-10531: ASCII Chars >127 not handled correctly if put at the begin or the end of the config item |
ASTERISK-10532: Bad DNS lookups on ENUMLOOKUP - various problems |
ASTERISK-10533: CLI: No more connections allowed |
ASTERISK-10534: gosubif doesn't allow empty true condition label similar to gotoif |
ASTERISK-10535: [patch] Issue with IMAP compilation |
ASTERISK-10536: new app_queue crashes |
ASTERISK-10537: revision 85764 introduces crash |
ASTERISK-10538: Add fractional timeouts and default timeout variable values to Read |
ASTERISK-10539: [PATCH] [sound] Add some functional to SayPosition in App_Queue |
ASTERISK-10540: Doesn't log TRANSFER in dynamic agent login |
ASTERISK-10541: [FR] - Respond with an error to SIP OPTIONS if there is an active shutdown |
ASTERISK-10542: [patch] Issues with threadstorage.c |
ASTERISK-10543: crash on vmcount |
ASTERISK-10544: Asterisk 1.4.13 segfaults at least once daily |
ASTERISK-10545: Read timeouts of less than 1ms result in using the default (6s) |
ASTERISK-10546: [patch] more NEW_CLI conversions |
ASTERISK-10547: GMT offset not stored/reported correctly |
ASTERISK-10548: Presence & eyebeam |
ASTERISK-10549: Implement READSTATUS |
ASTERISK-10550: glob #include uses base path different from non-glob #include |
ASTERISK-10551: hints display 'unreachable' peers still as 'idle' |
ASTERISK-10552: Circular call distribution no longer works |
ASTERISK-10553: One way audio when Dial() has both L option and more than one extension to ring |
ASTERISK-10554: i extention not all times work |
ASTERISK-10555: SIP MESSAGE request provokes disconnect after a dozen of seconds |
ASTERISK-10556: No ring tone is heard when calling a channel after the calling channel has been answered |
ASTERISK-10557: 500 ms delay on answer introduced channel locking issue ( issue 0008834) |
ASTERISK-10558: [patch] USB radio dependencies not properly passed to menuselect |
ASTERISK-10559: [patch] allow chan_usbradio to build under trunk with dev mode enabled |
ASTERISK-10560: add 'cdr' event to AMI |
ASTERISK-10561: Features (from features.conf) not available if call was originated by manager API or call file |
ASTERISK-10562: [patch] zap restart fails to generate channels |
ASTERISK-10563: patch for 10979 breaks IAX RSA auth |
ASTERISK-10564: Call Transfers core dump asterisk |
ASTERISK-10565: sip_poke_noanswer doesn't set hint state to unavailable |
ASTERISK-10566: False detection of USERSTOPPED status |
ASTERISK-10567: Calls Looping back into call groups cause confusion or overload. |
ASTERISK-10568: Unavailable peer not show in BLF of Thomson and Grandstream phones |
ASTERISK-10569: Asterisk segfaults on Playback of mp3 |
ASTERISK-10570: During transfers the local domain check fails if SIP phones provide a :portnr in their REFER message. |
ASTERISK-10571: Asterisk ignores port in register directive |
ASTERISK-10572: Added sample asterisk options to asterisk.conf |
ASTERISK-10573: Crash at random through the day. The box is Sip+H323 only |
ASTERISK-10574: Segfault in app_voicemail, inside c-client |
ASTERISK-10575: Record on sip trunk does not maintian voice codec data rate |
ASTERISK-10576: Crush at unknown place |
ASTERISK-10577: hint to monitor meetme using ael syntax |
ASTERISK-10578: change NEW_CLI to AST_CLI |
ASTERISK-10579: [patch] [sound] Add user/admin menu options to extend a RealTime scheduled conference |
ASTERISK-10580: CLI tab completion broken on "dundi show peer" |
ASTERISK-10581: [patch] update addons to use NEW_CLI API (and change ooh323 debug command) |
ASTERISK-10582: get_header_by_tagI() does not remove trailing <CR> |
ASTERISK-10583: [crash] FreeBSD: Crash if compile with option DEBUG_THREADS DONT_OPTIMIZE MALLOC_DEBUG |
ASTERISK-10584: compiling with MTX_PROFILE broken |
ASTERISK-10585: Deadlock.. i tjink in app_queue |
ASTERISK-10586: [patch] Solaris Fixes |
ASTERISK-10587: music on hold ends before new user enters |
ASTERISK-10588: implementation of application/dtmf for SIP INFO |
ASTERISK-10589: dtmf verbose |
ASTERISK-10590: [patch] Memory leak? |
ASTERISK-10591: Passing a NULL value back from IF() to Set() on 64-bit crashes Asterisk |
ASTERISK-10592: [patch] app_playback CLI command registered multiple times and formatting fixes |
ASTERISK-10593: [patch] Register to the SIP server with domain name |
ASTERISK-10594: Cannot detect correct extension number after several call-transfer. |
ASTERISK-10595: Asterisk doesn't recognize a "408 Request Timeout" |
ASTERISK-10596: Segfault in channel_spy() |
ASTERISK-10597: Segfault in ast_channel_spy_add() / AST_LIST_INSERT_TAIL |
ASTERISK-10598: Segfault in channel_spy() / __ast_pthread_mutex_unlock |
ASTERISK-10599: Asterisk-1.4.x :: Chan_H323 with Codec-g723 :: Metalic Sound |
ASTERISK-10600: Peers become unreachable when 'sip set debug' is enabled and reachable again when 'sip set debug off' |
ASTERISK-10601: Segfault when an agent login |
ASTERISK-10602: Asterisk SIP Connections to systems that support t38 fax detection may fail |
ASTERISK-10603: Crashes at random |
ASTERISK-10604: Devstate does not seem to work with SIP phones |
ASTERISK-10605: cdr_sqlite3_custom backend error when logging to db |
ASTERISK-10606: No audio on calls into queues with non-persistent agents |
ASTERISK-10607: [patch] Parked call timeout features |
ASTERISK-10608: Set CHANNEL(language)=es not changing spanish digit path |
ASTERISK-10609: G729 Codec does not load |
ASTERISK-10610: GROUP_COUNT() returns 0 |
ASTERISK-10611: Patch CLI app_meetme.c - Provides "meetme concise" |
ASTERISK-10612: hidecalleridname parameter don't work |
ASTERISK-10613: SIP channel stops processing calls, but no apparent deadlock |
ASTERISK-10614: Asterisk 1.4.13 stock segfault on pthread_mutex_lock |
ASTERISK-10615: [patch] Empty voicemail message file causes call into voicemail to disconnect |
ASTERISK-10616: asterisk segfault |
ASTERISK-10617: Hint stuck on hold after attended transfer with notifyhold=yes in sip.conf |
ASTERISK-10618: avoid nonexistent context warnings when loading AEL |
ASTERISK-10619: Add Theora Video to the list of codecs |
ASTERISK-10620: IMAP client fails with: IMAP Error: Unable to create selectable TCP socket (1024 >= 1024) |
ASTERISK-10621: Asterisk does pickup digits during call from CALLEE (feature.conf) |
ASTERISK-10622: Using func_curl in the [globals] section causes a segfault |
ASTERISK-10623: autopark strange behaviour if canreinvite=update,nonat |
ASTERISK-10624: [patch] Add MYSQL_OPT_RECONNECT for mysql client version |
ASTERISK-10625: CDR Created incorrectly on Transfer of outgoing call |
ASTERISK-10626: Segfault in strlen from ast_dynamic_str_thread_build_va __iax2_poke_noanswer |
ASTERISK-10627: asterisk releases progressively breaking queues |
ASTERISK-10628: [patch] Modify the return values on load_module(). |
ASTERISK-10629: SIP Registry Status: Unreachable |
ASTERISK-10630: chan_sip can't handle Cirpack KeepAlive Packets |
ASTERISK-10631: [patch] Add autoservice to all functions which might delay |
ASTERISK-10632: deadlock in iax |
ASTERISK-10633: Greater than 256 New messages crashes Asterisk |
ASTERISK-10634: ast_unescape_semicolon causes SIP NOTIFY to loop for snom-check-cfg and snom-reboot |
ASTERISK-10635: 'Fac_RESULT' and 'Fac_ERROR' undeclared |
ASTERISK-10636: [patch] func_cut.c and func_realtime.c no longer compile in dev mode |
ASTERISK-10637: [patch] Trivial duplicate #include entry |
ASTERISK-10638: Sent time on voicemail notification emails incorrect after upgrade to 1.4.13 |
ASTERISK-10639: [patch] to better debug main DSP busydetect function |
ASTERISK-10640: dialplan save not handling ;'s correctly |
ASTERISK-10641: Silence detector not working |
ASTERISK-10642: progressinband=yes send 180 and 183 together |
ASTERISK-10643: RaiseException() wrapped by ExecIf() causes segfault |
ASTERISK-10644: RaiseException() does not jump to 'e' extension |
ASTERISK-10645: [patch] Solaris fixes for editline compilation |
ASTERISK-10646: Followme does not delete temporary voice files |
ASTERISK-10647: [patch] *BSD mutex lock issue |
ASTERISK-10648: Make it obvious you can get the value of RaiseException(<reason>) from ${EXCEPTION(<type>)} |
ASTERISK-10649: Polycom Phones & Hints |
ASTERISK-10650: [patch] Make peer lookup more exact |
ASTERISK-10651: Some sound files have zero length |
ASTERISK-10652: Parkcall Does not allow to Re-park |
ASTERISK-10653: [branch] Implement asterisk CLI permissions. |
ASTERISK-10654: [patch] Check before trying to free in utils.c |
ASTERISK-10655: Asterisk fails to load func_curl.so |
ASTERISK-10656: hint does now work with the calling SIP channel |
ASTERISK-10657: Set CHANNELFORWARD variable to allow more control over setting CALLERID(num) on calls forwarded by 302 "Moved Temporarily" |
ASTERISK-10658: [patch] Memory leak on chan_gtalk.c |
ASTERISK-10659: [patch] Memory leak on res_jabber.c |
ASTERISK-10660: [patch] Memory lead on chan_jingle.c |
ASTERISK-10661: app_voicemail.c's imapfolder option is not documented |
ASTERISK-10662: Make the EXCEPTION() application use 'reason' instead of 'type' to get the error reason |
ASTERISK-10663: [patch] Use ast_free() instead of free() |
ASTERISK-10664: [patch] Memory leak on function odbc_log() if occurs an error condition. |
ASTERISK-10665: [patch] Fix for clean project in devmode |
ASTERISK-10666: [patch] use ast_free() instead of free(). |
ASTERISK-10667: [patch] pbx_lua.so: a lua pbx switch that allows dialplans written in pure lua |
ASTERISK-10668: [PATCH] Better handling of temporary channel name. Fixes FOP. |
ASTERISK-10669: Solaris build issues |
ASTERISK-10670: [patch] Fix small memory leak in config file proccessing when there are included files |
ASTERISK-10671: [patch] Improved support of QoS in channels using RTP |
ASTERISK-10672: 'h' extension is broken in trunk |
ASTERISK-10673: SayDigits seems to be broken in current trunk |
ASTERISK-10674: I need the uniqueid to show un core show channels concise |
ASTERISK-10675: Asterisk 1.4.13 Produces mutex errors and too many files open error |
ASTERISK-10676: issue 0011146 has not been fixed in trunk |
ASTERISK-10677: IMAP: Mailbox does not exist |
ASTERISK-10678: Over quota errors ignored |
ASTERISK-10679: Asterisk does not retry a call on receving a 3XX response |
ASTERISK-10680: callback failed on atxfer from members of queue |
ASTERISK-10681: Asterisk does not send a provisional response at every minute |
ASTERISK-10682: [patch] chan_unistm.c Memory leak & Deadlock & Crash |
ASTERISK-10683: pbx_builtin_setvar_helper crash |
ASTERISK-10684: CDR RECORD CAN'T BE ADDED (REGRESSION) |
ASTERISK-10685: New application app_pickupextn.c |
ASTERISK-10686: Added variable "agi_asteriskthreadid" to AGI |
ASTERISK-10687: [patch] free config structure while exiting on error. |
ASTERISK-10688: incoming audio lag |
ASTERISK-10689: Deadlock in channel.c |
ASTERISK-10690: [patch] Fixed not probable memory leak but possible on chan_zap.c |
ASTERISK-10691: [patch] Substitute the pipe with the comma on the applications documentation. |
ASTERISK-10692: Activate general jitterbuffer for Unistim |
ASTERISK-10693: unable to perform attended transfer of incoming call from mISDN |
ASTERISK-10694: Asterisk core not multitreading sip-to-sip calls |
ASTERISK-10695: Add option to ResetCDR allowing users to re-enable CDR (only) |
ASTERISK-10696: [patch] Audit of 'core show application <foo>' and try to make docs more consistant |
ASTERISK-10697: Allow direct RTP in "Dial" with "tT" options if phones are "canreinvite=yes" and "dtmfmode=info" |
ASTERISK-10698: MoH classes init failed |
ASTERISK-10699: Crash related to ast_module_user_remove |
ASTERISK-10700: WaitExten hangs up channel when first digit is entered |
ASTERISK-10701: "reload" command causes crash in Mac OSX Leopard 10.5 |
ASTERISK-10702: [patch] Prevent a LOG_WARNING while loading app_playback |
ASTERISK-10703: [patch] Prevent extensions ael WARNING while loading extensions.conf.sample |
ASTERISK-10704: fundamental (?) autoservice problem |
ASTERISK-10705: "core show channels verbose" small simple bug |
ASTERISK-10706: call limits not work as expected (limitonpeer, busy-level) |
ASTERISK-10707: Manager API hangs on "Command: show channels" |
ASTERISK-10708: Segfault on Action: Command / Command: core show channels concise |
ASTERISK-10709: menuselect is broken in revision 220 |
ASTERISK-10710: call routing based on caller-id fails to match |
ASTERISK-10711: Crash in chan_sip |
ASTERISK-10712: Time sent header is incorrect when sending voicemails via email |
ASTERISK-10713: chan_zap causing reset on E1 and eventually crashed asterisk |
ASTERISK-10714: DTMF minimal duration and Recommendation Q.24 |
ASTERISK-10715: MIX monitor file name and Agent Name that pickup call |
ASTERISK-10716: IAX truncking works on one side only |
ASTERISK-10717: [patch] Avoid asterisk WARNINGS while using the configs files in trunk configs directory |
ASTERISK-10718: RealTime MusicOnHold |
ASTERISK-10719: ff and rew buttons for control stream file do not work. |
ASTERISK-10720: chan_h323.c needs to be fixed to match trunk |
ASTERISK-10721: Problem when using NAT and Subscriptions |
ASTERISK-10722: Problem when using NAT and Subscriptions |
ASTERISK-10723: [patch] Add check_hangup() method to pbx_lua |
ASTERISK-10724: minor typo bug in ast_say_date_with_format_fr() |
ASTERISK-10725: when IMAP storage is enabled, a duplicate "regular" email is sent when no email account is specified |
ASTERISK-10726: Issues while cross compiling asterisk-addons |
ASTERISK-10727: Crush if DO_CRUSH |
ASTERISK-10728: DEBUG_THREADLOCALS: lock in main/threadstorage.c must be untracked? |
ASTERISK-10729: [patch] Some deadlocks while loading config |
ASTERISK-10730: [patch] Change free() to ast_free() |
ASTERISK-10731: SIP reload for large config halts SIP Processing |
ASTERISK-10732: [patch] Allow dialplan to set prefix for SIP Call ID |
ASTERISK-10733: [patch] expose zap DND mode to the dialplan |
ASTERISK-10734: [patch] Trivial: Use ast_free() insted of free() |
ASTERISK-10735: Restart of MOH doesn't work |
ASTERISK-10736: extensions.conf looks for a [default] context, even if not defined in zapata.conf |
ASTERISK-10737: The BLINDTRANSFER variable is not populated when call transfered via SIP 302 |
ASTERISK-10738: DLCX verb not recgonized |
ASTERISK-10739: trunk compilation error on FreeBSD (chan_unistim.c, hashtest) |
ASTERISK-10740: [patch] Avoid including not needed header files or already included. |
ASTERISK-10741: Email notification of voicemail segfaults Asterisk |
ASTERISK-10742: [patch] Error while linking with DEBUG_THREADLOCALS enabled |
ASTERISK-10743: [patch] Doxygen fixes for various files |
ASTERISK-10744: [patch] Create doxygen for hashtabs |
ASTERISK-10745: [patch] add 'concise' option to sip show channels |
ASTERISK-10746: URI direct dialing to target domain : call rejected if source extension exist in the destination dialplan. |
ASTERISK-10747: [patch] ast_cdr_free: CDR on channel 'SIP/02571-09174500' not posted |
ASTERISK-10748: crash at __ast_pthread_mutex_lock &pkt->owner->lock can not access |
ASTERISK-10749: bugs in udptl.c |
ASTERISK-10750: find lock |
ASTERISK-10751: Asterisk MUST NOT update Route-Set during in-dialog messages |
ASTERISK-10752: [patch] Many retransmits when chan_sip generates multiple outstanding requests |
ASTERISK-10753: SIP channel crashes on new call |
ASTERISK-10754: silence threshold put to configuration |
ASTERISK-10755: [patch] WaitForNoise added to accomplish WaitForSilence |
ASTERISK-10756: Parking -- execution does continue at next priority when requested parking extension is in use. |
ASTERISK-10757: Compilation of chan_iax fails on Fedora 8 |
ASTERISK-10758: [patch] cleanup in t38 structure initialization |
ASTERISK-10759: SRV lookups broken in SIP |
ASTERISK-10760: [patch] Asterisk-1.4.13 :: chan_h323 :: callerid(name) / h323id not sent during setup message |
ASTERISK-10761: Error while launching /etc/init.d/asterisk start in non-bash shell |
ASTERISK-10762: Distortion in Playback of .gsm files over non-GSM channel |
ASTERISK-10763: ast_variable_new() typo |
ASTERISK-10764: Asterisk unable to handle Multple Authorization Headers |
ASTERISK-10765: Answering local channel does not remove it from call path |
ASTERISK-10766: Ending recording with DTMF cuts off end of stream too early |
ASTERISK-10767: [patch] Add CallerIDNum: and CallerIDName: to Event: Hangup output |
ASTERISK-10768: Parking into already taken parking slot leaves stalls channels and partially blocks asterisk (fix to issue 11237 is bad) |
ASTERISK-10769: Blowup after one-two hours with Trunk |
ASTERISK-10770: Blowup after one-two hours with Trunk |
ASTERISK-10771: Typo in UPGRADE.txt |
ASTERISK-10772: crash ast_queue_hangup (chan=0x0) |
ASTERISK-10773: [patch] missing locks while calling astman_send_error() |
ASTERISK-10774: [patch] Deadlock in find_session(unsigned long ident) |
ASTERISK-10775: sip stops workking because of non rtp ports available |
ASTERISK-10776: Jitterbuffer does not work well with injected sound |
ASTERISK-10777: chan_h323 with H323Plus |
ASTERISK-10778: [patch] chan_h323 with H323Plus for TRUNK (SVN rev. 89183) |
ASTERISK-10779: Asterisk crash on non-responsive gateway |
ASTERISK-10780: [patch] HOLD notification for Polycom phones |
ASTERISK-10781: Asterisk doesn't reply well to in-dialog OPTIONS in pedantic mode |
ASTERISK-10782: [patch] Reuse code in hashtab.c |
ASTERISK-10783: When invalid IP address is specified chan_iax2 crashes. |
ASTERISK-10784: putnopvut is a bug |
ASTERISK-10785: [patch] Context tracing for channels |
ASTERISK-10786: chan_misdn don't compile in trunk because lock renamed |
ASTERISK-10787: [patch] ast_tvdiff_us |
ASTERISK-10788: [patch] QUEUE_MEMBER_COUNT doesn't load realtime queue members |
ASTERISK-10789: possible memory leak in asp_dsp_process |
ASTERISK-10790: On certain deadlocks, running core show locks segfaults asterisk and yields no lock info |
ASTERISK-10791: [patch] sysinfo dialplan function |
ASTERISK-10792: this patch makes transcoding smarter, |
ASTERISK-10793: [patch] trunk: rwlock tracking support (tracking and untracking static rwlock) |
ASTERISK-10794: [patch] add some constants to chan_zap |
ASTERISK-10795: [patch] Check if config file changed before reloading the configuration |
ASTERISK-10796: [patch] Execute AGI from the CLI and the manager interface |
ASTERISK-10797: Agent transfering cal via SIP transfer gets logged out |
ASTERISK-10798: [patch] Asterisk segfaults while doing a 'module reload'. |
ASTERISK-10799: [patch] usage of unitialized parameters in Monitor |
ASTERISK-10800: Failed compile pbx_ael on solaris |
ASTERISK-10801: [patch] Convert warnings to debug when using say.conf |
ASTERISK-10802: Crash with IAX2 incoming call: Bad magic number |
ASTERISK-10803: Asterisk crashed on reloading |
ASTERISK-10804: Addons build problems |
ASTERISK-10805: Unlock not locked. |
ASTERISK-10806: commit 8938 should be reverted |
ASTERISK-10807: Wrong parsing of application arguments Mysql Fetch |
ASTERISK-10808: [patch] find_context_locked() must return with conlock held |
ASTERISK-10809: [patch] Trivial: When logging, use features.conf insted of parking.conf |
ASTERISK-10810: astobj2.h needs size_t |
ASTERISK-10811: network.h needs compiler.h |
ASTERISK-10812: abstract_jb.h needs stdio.h |
ASTERISK-10813: frame.h needs stdint.h |
ASTERISK-10814: utils.h needs string.h |
ASTERISK-10815: utils.h needs stdarg.h |
ASTERISK-10816: Unexplained crash with very low volume |
ASTERISK-10817: blah blah test |
ASTERISK-10818: problems with counting call limits |
ASTERISK-10819: use comma in SIPPEER() |
ASTERISK-10820: [patch] Include action: getcategory for the GUI |
ASTERISK-10821: use 'busylevel' consistently |
ASTERISK-10822: Picking the wrong extension |
ASTERISK-10823: [patch] Trivial: Use ast_free() insted of free() |
ASTERISK-10824: Impliment CallForwardAll |
ASTERISK-10825: h323 does not compile in latest Trunk |
ASTERISK-10826: [patch] CLI command 'sip show history' parameter is a <call-id> not a <channel> |
ASTERISK-10827: [patch] Solaris build with editline |
ASTERISK-10828: [patch] Prevent an asterisk crash if we do a 'module unload app_dial.so' |
ASTERISK-10829: [patch] Solaris fixes to compile |
ASTERISK-10830: [patch] Add a check to the 'core show translation' function and fixed a text usage error. |
ASTERISK-10831: Fix for buil unistim in dev_mode (array subscript is above array bounds) |
ASTERISK-10832: When my gateway gets a new ip address, sip can not be re-registered (wrong "nonce"?) |
ASTERISK-10833: Unable to forward voice frame |
ASTERISK-10834: [patch] ooh323 does not compile with latest trunk |
ASTERISK-10835: CDR(billsec) return 0 in some scenarios |
ASTERISK-10836: Respect the original "From" header in "Dial" application |
ASTERISK-10837: ast_hint_state_changed locks conlock / hints in wrong order when monitoring SIP channels |
ASTERISK-10838: Errors in queues-with-callback-members.txt |
ASTERISK-10839: Issue in recordthread |
ASTERISK-10840: asterisk modifying the in-dialogue route set which is a violation of RFC3261 |
ASTERISK-10841: Asterisk hangs on reload, console not responsive |
ASTERISK-10842: [patch] chan_local does not propogate cause codes |
ASTERISK-10843: AEL macro argument variables aren't properly quoted when set |
ASTERISK-10844: [patch] CHANNELS dialplan function, get channel list in the dialplan |
ASTERISK-10845: [patch] remove obsolete code from dsp.c |
ASTERISK-10846: revision 89461 Problems with loader.c |
ASTERISK-10847: [patch] Trivial: Missing ast_module_user_remove() in res_features.c |
ASTERISK-10848: core show locks crush |
ASTERISK-10849: Hook-Flash behaviour results in echo-cancellation disabled on channel during call. |
ASTERISK-10850: ast_random() has a small bug |
ASTERISK-10851: AGI manager events |
ASTERISK-10852: "languageprefix=yes" doesn't work |
ASTERISK-10853: No audio midcall |
ASTERISK-10854: [patch] Allow channel unique IDs when searching for channels by name |
ASTERISK-10855: MTX_PROFILE broken again ael_main.c |
ASTERISK-10856: [patch] Implement call park |
ASTERISK-10857: [patch] Remove privacy.conf and minor code cleanup. |
ASTERISK-10858: [patch] Don't try say number after hungup (new style) |
ASTERISK-10859: [patch] dialplan remove extension make coredump if asterisk compiled with MALLOC_DEBUG options |
ASTERISK-10860: RTP session ID is negative half the time on x86_64 |
ASTERISK-10861: [patch] Deprecate SIPPEER()/IAXPEER() and move functionality into CHANNEL() |
ASTERISK-10862: Crash on calls AST_LIST_REMOVE_HEAD Macro |
ASTERISK-10863: [patch] Trivial: Missing ast_frfree() in res_adsi.c |
ASTERISK-10864: Problem with Like query on MS SQL |
ASTERISK-10865: Can not change Call-Waiting tone duration and frequency |
ASTERISK-10866: [patch] Missing sched_context_destroy() in ast_channel_alloc() error condition. |
ASTERISK-10867: [patch] Promote more reuse in hashtab.c |
ASTERISK-10868: filestream is not closed before executing commands |
ASTERISK-10869: [patch] Trivial: Use ast_free() insted of free() |
ASTERISK-10870: setting voicemail greeting when using IMAP backend causes zero byte messages to be left |
ASTERISK-10871: backport ast_debug to 1.4 |
ASTERISK-10872: [patch] serving multiple Realms with one Asterisk |
ASTERISK-10873: [patch] Error compiling chan_h323 with DEBUG_THREADS |
ASTERISK-10874: Picking the wrong extension with _21x. |
ASTERISK-10875: ESCAPE clause in first parameter not escaped properly |
ASTERISK-10876: [patch] Add 'voicemail reload' CLI command |
ASTERISK-10877: `sip show channels' does not display properly the codec in use for audio and video calls |
ASTERISK-10878: [patch] Avoid asterisk crash when unloading module app_meetme |
ASTERISK-10879: chan_mobile does not recognize dtmf together with Authenticate or DISA |
ASTERISK-10880: Asterisk crash |
ASTERISK-10881: Add Extension without information produced NO LOGIN ANYMORE |
ASTERISK-10882: People with a 2 letter surname cannot be looked up from the directory |
ASTERISK-10883: Dial option L(limit in ms) is not working |
ASTERISK-10884: crash at manager.c pointer error |
ASTERISK-10885: Asterisk crash when executing SQLPREPARE statememt at cdr_odbc.c |
ASTERISK-10886: app_controlplayback used with option o (restart playback at position) craches Asterisk |
ASTERISK-10887: [patch] Codec negotiation results in asterisk sending unsupported codec |
ASTERISK-10888: added a return variable to app_controlplayback that reports the key used to stop playback. |
ASTERISK-10889: [patch] Update documentation that both 'num' and 'number' are valid for use in CALLERID() |
ASTERISK-10890: 1.4.14 breaks cdr posting |
ASTERISK-10891: having same voicemail pin as voicemail password should force user to change their password, but it does not |
ASTERISK-10892: [patch] Update examples to use CALLERID(num) instead of CALLERID(number) |
ASTERISK-10893: Dial option G does not handle labels under some conditions |
ASTERISK-10894: zaptel handel leak in meetme conference |
ASTERISK-10895: instalation problam |
ASTERISK-10896: Option eventwhencalled seems not working properly. |
ASTERISK-10897: Asterisk 1.4.14 and load average. |
ASTERISK-10898: Loading of res_config_pgsql will crash on dbhost/dbsock combination |
ASTERISK-10899: voicemail directory is not created until user is left a message |
ASTERISK-10900: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever. |
ASTERISK-10901: [patch] Trivial: Replace free() with ast_free() to match up with ast_calloc() / ast_malloc() |
ASTERISK-10902: caller_chan_id/callee_chan_id does not contain correct info when call is xfered |
ASTERISK-10903: Using Net-SNMP (RPM) to compile Asterisk with SNMP on CentOS 4 not possible. |
ASTERISK-10904: [patch] Trivial: Replace free() with ast_free() to match up with ast_calloc() / ast_malloc() |
ASTERISK-10905: [patch] Add missing includes in SVN Trunk |
ASTERISK-10906: [patch] hide CLI commands starting with '_' |
ASTERISK-10907: Few unfortunate misprint in log messages |
ASTERISK-10908: Dont play video on console dial |
ASTERISK-10909: [patch] Send Asterisk version to AGI |
ASTERISK-10910: Crash with signal 6 on Channel Hangup |
ASTERISK-10911: Resetting the SEQ number back to 0 without sending a new INVITE SSRC |
ASTERISK-10912: Variable identifier not compatible with C++ |
ASTERISK-10913: Interdigit timeout is half time of the defined time |
ASTERISK-10914: [patch] Remove old CLI style. |
ASTERISK-10915: SMDI MWI configuration on a per-user basis |
ASTERISK-10916: [patch] free the returned data after a ast_load_realtime() |
ASTERISK-10917: New voicemail count incorrect when using IMAP storage and delete=yes flag |
ASTERISK-10918: IAX crashing Asterisk |
ASTERISK-10919: Buffer overflow when maxmsg not used for IMAP storage users |
ASTERISK-10920: Temporary greeting prompt could give more immediate feedback of state (greeting in use) |
ASTERISK-10921: Reopen bug 0009650 [No ringing heard on unattanded transfer] |
ASTERISK-10922: [patch] Default penalty for member added using QueueAdd |
ASTERISK-10923: [patch] Trivial: Replace free() with ast_free() to match up with ast_calloc() |
ASTERISK-10924: Strange freezing of the Manager and Asterisk Console |
ASTERISK-10925: [patch] Move loading users from authenticate() to __init_manager() |
ASTERISK-10926: Better quota handling with IMAP storage |
ASTERISK-10927: [patch] Do not log debug messages unless debug is enabled |
ASTERISK-10928: chanisavail always returns the same status |
ASTERISK-10929: DTMF buffer not purged if the dialing timeout |
ASTERISK-10930: Saving to IMAP folder (other than INBOX) not working with Cyrus IMAP |
ASTERISK-10931: Calls barf and could crash on onelegged transfer |
ASTERISK-10932: MeetMe conferences don't forward DTMF from SIP clients |
ASTERISK-10933: [patch] Add new CLI command 'core show hint <exten>' and autocomplete for this command too. |
ASTERISK-10934: MySQL addons break latest asterisk 1.4.15 |
ASTERISK-10935: Asterisk 1.4.15 breaks format_mp3.c build |
ASTERISK-10936: Dailstatus says NOANSWER even if i pick the call |
ASTERISK-10937: Random seg faults... |
ASTERISK-10938: Asterisk crashes when using ODBC connected to Sybase |
ASTERISK-10939: incorrectly recognized codecs capabilities of windows mobile 6 device |
ASTERISK-10940: Sent RTP video packets have a timestamp based on a 8000 Hz clock instead of 90000 Hz when mark bit is on |
ASTERISK-10941: [patch] Small build issue on solaris |
ASTERISK-10942: [patch] Trivial: Replace free() with ast_free() to match up with ast_calloc() / ast_malloc() |
ASTERISK-10943: [patch] Sample conf update: Define a list of periodic announcements |
ASTERISK-10944: ENUMLOOKUP broken in 1.4.15 |
ASTERISK-10945: asterisk 1.4.14, both autoconf and configure on Ubuntu are broken |
ASTERISK-10946: app_queue crashes on startup |
ASTERISK-10947: Calling parkandannounce from the dial plan causes Asterisk to core dump |
ASTERISK-10948: Compiling Asterisk --with-imap support fails |
ASTERISK-10949: [patch] Avoid calling twice ast_var_full_name(). |
ASTERISK-10950: i applied chan_zap.c.diff patch i works fine Airtel in bangalore |
ASTERISK-10951: i applied chan_zap.c.diff patch it works Airtel line with some bug. but BSNL line not working in bangalore |
ASTERISK-10952: Good time fun crash when using Dial() |
ASTERISK-10953: duplicate values when rewriting templated config files |
ASTERISK-10954: Call hangup if send a DTMF during phone ringing in DTMF HYBRID MODE |
ASTERISK-10955: after a dialplan reload hints for parked calls are allways in use |
ASTERISK-10956: Memory Leak when running asterisk -rx "xxxxxx" |
ASTERISK-10957: Standard "free()" memory function replaced with "ast_free()" in Voicemail code |
ASTERISK-10958: asterisk permanently burden cpu |
ASTERISK-10959: SIP RTP crash |
ASTERISK-10960: [patch] Don't reload the configuration if the .conf hasn't change |
ASTERISK-10961: Change Makefile to match tex documentation |
ASTERISK-10962: FASTAGI disconnects all ssh sessions to PC when hangup occurs |
ASTERISK-10963: [patch] Close timing interface on error |
ASTERISK-10964: Asterisk crash in the middle of a "reload" command |
ASTERISK-10965: Configuration setting when i am behind nat |
ASTERISK-10966: Configuration setting when i am behind nat |
ASTERISK-10967: Asterisk crashed - App Queue receiving calls |
ASTERISK-10968: [patch] Free agent and waiting callers |
ASTERISK-10969: [patch] Remember to update the CHANGES file when a new feature is added. |
ASTERISK-10970: safe_asterisk fails to restart asterisk after crash |
ASTERISK-10971: Long usernames doesn't appear in manager or CLI |
ASTERISK-10972: Outgoing Calls |
ASTERISK-10973: Get storage api to compile |
ASTERISK-10974: [patch] h extension not executing, zap channel getting stuck |
ASTERISK-10975: IF() function crashes when using invalid syntax |
ASTERISK-10976: Cleanup in SayCountPL |
ASTERISK-10977: Left-over line in http.conf |
ASTERISK-10978: Asterisk 1.2.25 crashes with undefined symbol in cdr_pgsql |
ASTERISK-10979: Asterisk hangup calls if two identical invites are received |
ASTERISK-10980: Asterisk replies with a 488 to a Re-Invite with no SDP |
ASTERISK-10981: asterisk: symbol lookup error: /usr/lib/asterisk/modules/cdr_odbc.so: undefined symbol: ast_odbc_request_obj |
ASTERISK-10982: VoiceMailMain core at ast_adsi_transmit_message_full (res_adsi.c:358) |
ASTERISK-10983: [patch] Add manager action 'CoreShowChannels' |
ASTERISK-10984: ZTTEST drops to 96% with only 4-5 Calls on Circuit |
ASTERISK-10985: Asterisk rejects legitimate G.729a call with 488 Not Acceptable Here |
ASTERISK-10986: Issue on solaris sparc with compilation of aelparse in utils/ |
ASTERISK-10987: Alignment issues building on solaris sparc |
ASTERISK-10988: Core dump <tzafrir> Astrisk passed a bad pointer to libc |
ASTERISK-10989: [patch] New application to sleep until a given UNIX epoch |
ASTERISK-10990: [patch] During call setup signalling asterisk does not offer telephone-event (rfc2833) anymore |
ASTERISK-10991: [patch] Spelling mistake and extra indent in chan_sip.c |
ASTERISK-10992: RTP timestamp skewed after return from hold. |
ASTERISK-10993: [patch] fix documentation for pbx_lua |
ASTERISK-10994: Asterisk providing SIP proxy with incorrect routing information |
ASTERISK-10995: The [VM_DATE] never gets translated into the locales. |
ASTERISK-10996: [patch] extra voicemail information via manager interface |
ASTERISK-10997: RINGNOANSWER queue_log without a member name |
ASTERISK-10998: Crush if i try call to queue that not have memebers |
ASTERISK-10999: some diferences in AMI events asterisk1.2 <=> astreisk |